similar to: Asterisk Cnfiguration requirements?

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10]

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was

Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

Hi, I am not new to asterisk but this is the first time im using iax protocol. I have always used sip before, but i have heard its better to use iax if i want communication between 2 asterisk servers. I just registered asterisk server1 with asterisk server2 and tried to call server1 from 2 but the call does not pass thru. i dont see any messages on the recieving asterisk cli but the caller

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

Hi all, There is a parameter called "nonce" included in every register request that a UA sends to asterisk. I have read sip debug a lot and only found out that the "nonce" parameter value which is used in register request was generated by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). <--- Transmitting (NAT) to

hi all, actually i have partially connected the 2 servers but there is a problem. 2 servers A and B server A forwards call to server B without any problem but when i try to forward call from server B to A, server shows the following error on the cli WARNING[7751]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the